Jean
If you moved the exten => _. Lines to the bottom of the context then you
should like be able to get away from having to have two separate contexts. I
use that method quiet often, but was in a hurry to get you a response and did
not think remember that nuance.
I will have to try this
> Hi All,
>
> I've setup a Digium G100 VoIP gateway to replace an internal PCI VoIP
> card in our Asterisk PBX. When using the VoIP card the callerid entries
> listed in sip.conf were displayed when calling someone over the PSTN.
> Now, however, though the gateway it just displays the
Hi All,
I've setup a Digium G100 VoIP gateway to replace an internal PCI VoIP
card in our Asterisk PBX. When using the VoIP card the callerid entries
listed in sip.conf were displayed when calling someone over the PSTN.
Now, however, though the gateway it just displays the default number
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Hi Bryant,
Thanks for your reply.
It didn't work immediately, I had to create a second context, or else it
was looping between the second and first line. This seems to work:
[earlydial] ; Test Early Dial
exten => _.,1,Set(l_Extension=${EXTEN})
Has anyone created any docker images I might be able to use on EC2 for
load testing an asterisk platform? I started an instance this morning
and was about to load sipp and other tools, and then thought surely
someone must have done this already. I'd like to hammer a platform we
have
Jean-Denis Girard
I have not used the Incomplete yet, but you might be able to do something like
this.
[earlydial]
exten => _.,1,Set(l_Extension = ${EXTEN})
exten => _.,n,Goto(${l_Extension},1)
exten => _.,n,Goto(noMatch,1)
exten => i,1,Goto(noMatch,1)
exten => noMatch,1,
Phillip
Check out the b and B options one of them should do what you want.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: "Saint Michael"
Dear friends:
Is there a way to execute a macro or sub-routine after we send the invite
before we receive anything like a 200 OK, 183, etc?
Philip
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New
on my own server
i want try jssip
https://github.com/versatica/JsSIP
it looks like a lot "livelier" than sipml5
any experience with jssip?
Dne 18.2.2016 v 16:01 Olivier napsal(a):
2016-02-18 15:42 GMT+01:00 Marek Červenka >:
my